Vodia PBX
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We get the basic on/off stuff working. However, there is a problem when the packets get too big (more than 1492) if the transport layer is UDP. This happens when the initial NOTIFY lists all buddies - then the UDP packet gets fragmented and the phone cannot use it. Switching to TCP transport layer seems to solve that problem.
Our problem is that in the beginning things look fine, then after some time the display gets kind of garbeled. That might relate to the firmware version that we are using. It seems to happen after the first few calls, but I think that was fixed in on of the latest firmware releases. Did not have the time to upgrade to one of these releases yet.
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I would start monitoring one or two extensions first, this avoids problems with too long messages. XML tends to become very long
So far I have seen only on/off states in the leds. The dialog-state based BLF is IMHO not able to provide features like pickup. If you have a packet trace that shows how to make the Polycom start blinking that would be very interesting.
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Sounds like a problem with NAT or the firewall to me...
Is there anything on http://wiki.pbxnsip.com/index.php/Troubles..._Trunk_Problems that we should add to the Wiki checklist?
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That sounds like there is something else running on this machine that is SIP-aware. Maybe set the PBX to start manually, reboot the system and check with netstat what processes are listening on port 5060. If there is something, make sure it does not fight for the same ports as the PBX...
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Did you check the Wiki on that topic? See http://wiki.pbxnsip.com/index.php/Microsoft_Exchange.
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What is strange here is that 300 seems to be an account, but the PBX does not find it and uses the dial plan to dial an external number. Do we have a problem of mixing up domains here?
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You can override these settings only locally on the phone (e.g. using the web browser). The PBX won't override them next time when the phone reboots and refreshes the provisioning data.
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Does the recording mechanism still apply or not?
It should. If it is not recorded, we are talking about a bug.
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The easest way to do this is to generate files with the standard PnP mechanism, let the PBX write them into the "generated" directory, move them into the tftp directory and then use them as a template.
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Ouch. Sounds like a bug.
Will be fixed in 2.1.3.
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Hunt groups don't call external numbers on the hunt stages, and they also don't follow extension redirects.
Having external numbers in a hunt group is asking for trouble. In FXO, many gateways will connect the call immediately (playing ringback tone or just comfort noise). And cell phones tend to get offline (driving in a tunnel or user goes to bed), then the mailbox picks up and connects the call immediately. And calling a busy PSTN line in many cases goes to some kind of IVR (mailbox, busy annoucement) and also connect the call immediately.
When the PBX calls a SIP extension, it can avoid all of that and have the hunt group doing something predictable and supportable.
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If I were technically capable, I think adding some SNOM action URL's to the PBXnSIP web server to do some cool stuff like this would be a real plus. The SNOM web - wiki speaks of this exact example, but we simply don't want to go about creating 1-off solutions.
The address book button goes to the PBX address book. At least something!
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The purpose of that SUBSCRIBE was to get information about the hook state of the phone. This should be obsolete in the latest and the greatest version (2.1.2), so in theory if that was the problem that problem should disappear by itself...
Any chance to get a core dump? By looking at the raw memory it should be possible to see what is occupying the memory...
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Well, it can be done. See http://wiki.pbxnsip.com/index.php/Snom. Just make sure that you are using version 7.1.28 or higher. There are images available at http://www.pbxnsip.com/protect/snom3[0267]0-7.1.28-SIP-f.bin, just in case that you don't find them on the Internet.
The monitoring of the DND button for other extensions might be a little challenge, though. Maybe this is something that we need to add...
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I am using the External Voice Mail system settings for Exchange. Does this defeat the Mailbox direct Dial Prefix?
Yes. In this case you should directly call the Exchange with a different prefix that you are using in the dial plan to call Exchange. For example, if you are using 999* in the dial plan to send calls to the Exchange trunk, then dial 999123 if 123 is your account.
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I think the best way to deal with this is to have a "personal hunt group" that is sitting in front of the extension. The only member in this hunt group is your personal extension. There you can define that you can send calls after hours directly to your voicemail (e.g. 8123 is you personal extension is 123 and the domain direct mailbox prefix is 8). People would call that hunt group from the auto attendant.
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We just made version 2.1.2 available (build 2292). The release notes are available from:
http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.2
If you want to use the new conference name recording feature, you need to reload the audio files (language and also the moh files). We updated them as well, you can just download the files and overwrite the existing audio files.
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When a hunt goup is calling the extension, it does not use the auto attendant to do this (the auto attendant also calls the cell phone). This was done by intention - if a hunt group would also call the cell users would not use the cell feature as they would get called for every group call. BTW the same is the case for call redirect on busy/timeout/always, the hunt group does not follow those redirections.
There is a trick that you can use to make the PBX call the cell phone all the time: Add a static registration with the SIP address of the gateway (e.g. sip:2121234567@ip-of-gateway). Then the PBX will send absolutely every call to the cell phone. But be careful with this feature: If you have a large hunt group, every call to that group will send a lot of new calls to your gateway.
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That can be done by saying "ring 99 agents every ten seconds" (unless you have more than 99 agents). See the "Call rate limitation", in this case "99/10".
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Well, you can always use a hunt group to address a user. The real extension number is "hidden", just used to really ring the extension when it is time to do so. If you want to route directly to the mailbox, you can put the mailbox prefix in from of the extension number.
The cell phone redirection feature has the day/nightmode that you are talking of. Maybe you can use the Exchange number to implement the behavior if you provision the cell phone with the Exchange account.
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Just updated http://wiki.pbxnsip.com/index.php/Auto_Att...ct_Destinations. You can use # behind the direct destination.
But better avoid that (if you can) by choosing extension number starting with 4-7.
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The secretary can override DND - see http://wiki.pbxnsip.com/index.php/Extension#Permissions.
BTW this is another reason to have the DND state on the PBX, not on the phone.
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Almost - after playing the prompt (always, with background music) it will start playing ringback to tell the caller that there is an agent running to his or her phone to grab the call.
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Well the purpose of the ringback tone is to indicate the caller to pay close attention now as probably one of the agents is running to the phone to grab the call. Music on hold says relax, this might take a while.
What you can do is record something in prompt 0 of the queue, maybe something like "welcome to xxx, we are talking your call very serious. Did you know we have a new product". Then the caller will hear MoH and will be happy that there is a agent ready for him right away.
Callers are usually delighted if they don't have to listen to MoH for too long...
Call Routing
in General Setup
Posted
Probably it is easier to look into the concrete system that talking abstract.